Reformat the WebRTC code base
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
index 6288aeb..273979b 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -33,25 +33,25 @@
class AudioDecoderFactory;
struct NetEqNetworkStatistics {
- uint16_t current_buffer_size_ms; // Current jitter buffer size in ms.
+ uint16_t current_buffer_size_ms; // Current jitter buffer size in ms.
uint16_t preferred_buffer_size_ms; // Target buffer size in ms.
- uint16_t jitter_peaks_found; // 1 if adding extra delay due to peaky
- // jitter; 0 otherwise.
- uint16_t packet_loss_rate; // Loss rate (network + late) in Q14.
- uint16_t expand_rate; // Fraction (of original stream) of synthesized
- // audio inserted through expansion (in Q14).
+ uint16_t jitter_peaks_found; // 1 if adding extra delay due to peaky
+ // jitter; 0 otherwise.
+ uint16_t packet_loss_rate; // Loss rate (network + late) in Q14.
+ uint16_t expand_rate; // Fraction (of original stream) of synthesized
+ // audio inserted through expansion (in Q14).
uint16_t speech_expand_rate; // Fraction (of original stream) of synthesized
// speech inserted through expansion (in Q14).
- uint16_t preemptive_rate; // Fraction of data inserted through pre-emptive
- // expansion (in Q14).
- uint16_t accelerate_rate; // Fraction of data removed through acceleration
- // (in Q14).
- uint16_t secondary_decoded_rate; // Fraction of data coming from FEC/RED
- // decoding (in Q14).
+ uint16_t preemptive_rate; // Fraction of data inserted through pre-emptive
+ // expansion (in Q14).
+ uint16_t accelerate_rate; // Fraction of data removed through acceleration
+ // (in Q14).
+ uint16_t secondary_decoded_rate; // Fraction of data coming from FEC/RED
+ // decoding (in Q14).
uint16_t secondary_discarded_rate; // Fraction of discarded FEC/RED data (in
// Q14).
- int32_t clockdrift_ppm; // Average clock-drift in parts-per-million
- // (positive or negative).
+ int32_t clockdrift_ppm; // Average clock-drift in parts-per-million
+ // (positive or negative).
size_t added_zero_samples; // Number of zero samples added in "off" mode.
// Statistics for packet waiting times, i.e., the time between a packet
// arrives until it is decoded.
@@ -104,11 +104,7 @@
absl::optional<AudioCodecPairId> codec_pair_id;
};
- enum ReturnCodes {
- kOK = 0,
- kFail = -1,
- kNotImplemented = -2
- };
+ enum ReturnCodes { kOK = 0, kFail = -1, kNotImplemented = -2 };
// Creates a new NetEq object, with parameters set in |config|. The |config|
// object will only have to be valid for the duration of the call to this